1. Field
One embodiment of the invention relates to a telephone system realizing voice communication via an Internet Protocol (IP) network, an exchange for use in the telephone system, and a transmission control method.
2. Description of the Related Art
A system which realizes voice communication by means of a best-effort network, such as the Internet, is well known. This kind of system is called a voice-over-IP (VoIP) or IP telephone system and is expected as a next-generation telephone system. Session Initiation Protocol (SIP) represents a protocol usable in this system. Providing a SIP processing function for a telephone set (hereinafter, referred to as an IP terminal) achieves various services unique to SIP. Of course, various IP terminals including conventional telephone sets having no SIP processing function may be connected to the SIP network.
The SIP network, since it is well suited to a local area network (LAN), is frequently configured as a private telephone network. In transmitting to an external network from the private network, a SIP message “INVITE” is transmitted to a SIP server, which handles an external network, through a SIP trunk of a private branch exchange (PBX).
If, at this time, the SIP server has failed, call control cannot be performed and a new communication link cannot be formed. To avoid this situation, the SIP server has a redundant configuration. If a non-response status continues after transmission of the SIP message to the SIP server, a caller accesses sequentially the next SIP server. However, if all the SIP servers have failed, the telephone system cannot transmit any SIP message from the SIP trunk. In this case, the telephone system has to switch to a detour transmission using another trunk (analog trunk, Integrated Services Digital Network [ISDN] trunk, etc.).
In the existing technique, a transaction timer default value of the SIP message (INVITE) is set to 32 seconds. If the caller cannot access all the SIP servers, the caller has to wait by the time length in which 32 seconds is multiplied by the number of SIP servers. Shortening the transaction timer value can make the time to start the detour transmission shorter, reducing how long the telephone system has to wait. To eliminate the waiting time, it is necessary for the PBX to recognize the status of the SIP trunk (or SIP server) before a transmission operation to allow the detour transmission to be performed immediately.
A method for checking statuses of SIP terminals from the SIP server has been disclosed (refer to Jpn. Pat. Appln. KOKAI Publication No. 2006-1660189). However, the technique of this reference document may not be applied to check statuses of SIP server or the SIP trunk from the SIP terminals. With reference to a result of registration that is a function of the SIP network, monitoring the status of the SIP server is a possible method. However, in recent years, the number of SIP networks having no registration functions has increased, so that the method of the above is not useful for a wide variety of purposes and lack of accuracy.
As mentioned above, the existing technique cannot determine the status of the SIP server by means of a general-purpose method before transmission. If a caller originates a call in a status in which the SIP server and the SIP trunk have failed, since the caller has to wait a long time until the detour transmission is established, some sort of countermeasure has been desired.